vb.net barcode generator Audio Codec An audio codec encodes the audio signal from the microphone in Software

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Audio Codec An audio codec encodes the audio signal from the microphone
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for transmission on the transmitting H.323 terminal and decodes the received audio code that is sent to the speaker on the receiving H.323 terminal. Because audio is the minimum service provided by the H.323 standard, all H.323 terminals must have at least one audio codec support, as specified in the ITU-T G.711 recommendation (audio coding at 64 Kbps). Additional audio codec recommendations such as G.722 (64, 56, and 48 Kbps), G.723.1 (5.3 and 6.3 Kbps), G.728 (16 Kbps), and G.729 (8 Kbps) may also be supported.
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Video Codec A video codec encodes video from the camera for transmission
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on the transmitting H.323 terminal and decodes the received video code
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Downloaded from Digital Engineering Library @ McGraw-Hill (www.digitalengineeringlibrary.com) Copyright 2004 The McGraw-Hill Companies. All rights reserved. Any use is subject to the Terms of Use as given at the website.
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Voice over Internet Protocol
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that is sent to the video display on the receiving H.323 terminal. Because H.323 specifies the support of video as optional, the support of video codecs is optional as well. However, any H.323 terminal providing video communications must support video encoding and decoding as specified in the ITU-T H.261 recommendation.
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H.225 Registration, Admission, and Status (RAS) RAS is the protocol between endpoints (terminals and gateways) and gatekeepers. RAS is used to perform registration, admission control, bandwidth changes, and status, and to disengage procedures between endpoints and gatekeepers. An RAS channel is used to exchange RAS messages. This signaling channel is opened between an endpoint and a gatekeeper prior to the establishment of any other channels. H.225 Call Signaling The H.225 call signaling is used to establish a con-
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nection between two H.323 endpoints. This is achieved by exchanging H.225 protocol messages on the call-signaling channel, which is opened between two H.323 endpoints or between an endpoint and the gatekeeper.
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H.245 Control Signaling H.245 control signaling is used to exchange end-
to-end control messages governing the operation of the H.323 endpoint. These control messages carry information related to the following: capabilities exchange, the opening and closing of logical channels used to carry media streams, flow-control messages, general commands, and indications.
Real-Time Transport Protocol (RTP) and H.323 RTP, a transport protocol, pro-
vides end-to-end delivery services of real-time audio and video. Whereas H.323 is used to transport data over IP-based networks, RTP is typically used to transport data via the UDP. RTP, together with UDP, provides transport-protocol functionality. RTP provides payload-type identification, sequence numbering, timestamping, and delivery monitoring. UDP provides multiplexing and checksum services. RTP can also be used with other transport protocols.
Real-Time Transport Control Protocol (RTCP) and H.323 RTCP is the coun-
terpart of RTP that provides control services. The primary function of RTCP is to provide feedback on the quality of the data distribution. Other RTCP functions include carrying a transport-level identifier for an RTP source, called a canonical name, which is used by receivers to synchronize audio and video.
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Voice over Internet Protocol
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SIP SIP is a text-based signaling protocol used for creating and controlling multimedia sessions with two or more participants. It is a client-server protocol transported over TCP. SIP can interwork with gateways that provide signaling protocols and media translations across dissimilar network segments such as PSTN to IP networks. SIP uses text-based messages, much like HTTP. SIP addressing is built around either a telephone number or a web host name. In the case of a web host name, the SIP address is based on a uniform resource locator (URL). The URL is translated into an IP address through a domain name server (DNS). SIP also negotiates the features and capabilities of the session at the time the session is established.5 SIP plays such a pivotal role in the evolution of VoIP and softswitching that it must be covered in the separate, following chapter. Gateway Control Protocols The most immediate attraction to VoIP is to save money on long-distance transport. To date, it has been impractical to route VoIP desktop to desktop, meaning interworking between PSTN and IP networks must be facilitated. This is done with a gateway. The two most applied gateways are the media gateway and the signaling gateway. Media gateways interconnect dissimilar networks. In this case, they connect the PSTN to IP networks. To do this successfully, they must intermediate both signaling and transport between the two dissimilar networks (PSTN and IP). Media gateways coordinate call control and status. Gateway control protocols are signaling protocols. MGCP MGCP is the protocol used to intermediate the Media Gateway Controller (MGC, also known as a call agent) and the media gateway. MGCP was developed by the IETF and details the commands and parameters that are passed between the MGC and the telephony gateway to be controlled. MGCP assumes a call control architecture where the call control intelligence is outside the gateways and is handled by external call control elements. The MGCP assumes that these call control elements, or call agents, will synchronize with each other to send coherent commands to the gateways under their control. MGCP is a master/slave protocol, where the gateways are expected to execute commands sent by the call agents. The purpose of MGCP is to send commands from the call agent to a media gateway (see 3 for descriptions of media gateways). MGCP defines both endpoints and connections. Endpoints are sources or sinks of
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